Steve Murphy
4d8932a6dc
Merged revisions 89622 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Steve Murphy
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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89513 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Luigi Rizzo
7e8835e0d7
remove another set of redundant #include "asterisk/options.h"
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Luigi Rizzo
a23c055c3d
move asterisk/paths.h outside asterisk.h and into those files
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who really need it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Luigi Rizzo
0595b5e2aa
include "logger.h" and errno.h from asterisk.h - usage shows that they
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were included almost everywhere.
Remove some of the instances.
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18 years ago
Luigi Rizzo
fdb7f7ba3d
Start untangling header inclusion in a way that does not affect
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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.).
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18 years ago
Russell Bryant
0df5e50e97
Use BEGIN_OPTIONS / END_OPTIONS to make the syntax highlighting in my editor happy
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
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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18 years ago
Matthew Fredrickson
a4be521c89
Make sure we propogate ANI2 to the outbound channel
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18 years ago
Tilghman Lesher
7adbd6bb16
Remove redundant includes (patch by snuffy) (Closes issue #10922 )
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18 years ago
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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18 years ago
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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18 years ago
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
18 years ago
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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18 years ago
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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18 years ago
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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18 years ago
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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18 years ago
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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18 years ago
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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18 years ago
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).
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18 years ago
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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18 years ago
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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18 years ago
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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18 years ago
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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18 years ago
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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18 years ago
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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18 years ago
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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18 years ago
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
18 years ago
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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18 years ago
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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18 years ago
Steve Murphy
2462d5ab4f
Cleaning up a small disaster I created earlier
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70607 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Steve Murphy
57526b35cc
As per 9228, now app_queue should have the proper machinery to do gosubs.
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18 years ago
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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18 years ago
Tilghman Lesher
704c756c4a
Merge work to make U(...) option work for Dial
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18 years ago
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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18 years ago
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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18 years ago
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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18 years ago
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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18 years ago
Tilghman Lesher
9d05ff8ed5
Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
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18 years ago
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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18 years ago
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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18 years ago
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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18 years ago
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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18 years ago
Russell Bryant
be874b92d3
Remove unused instances of unnamed enums.
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18 years ago
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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18 years ago
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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19 years ago
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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19 years ago
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54925 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54885 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
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
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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19 years ago
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
32cd307d6f
Merged revisions 53749 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
19 years ago
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
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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
19 years ago
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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19 years ago
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
19 years ago
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
19 years ago
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
19 years ago
Luigi Rizzo
9d4531d636
move a large block related to privacy handling to a separate function.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
359a553961
Merged revisions 48193 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48193 | kpfleming | 2006-12-01 17:37:28 -0600 (Fri, 01 Dec 2006) | 10 lines
Merged revisions 48192 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006) | 2 lines
if Dial() is going to send music-on-hold to the calling party, it has to send PROGRESS first to ensure that the reverse audio path has been setup first (BE-106)
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19 years ago
Luigi Rizzo
bbba4e5a20
better fix for the previous bug.
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In general this code needs a deep revision, because the body of
do_forward() deletes/overwrites the output channel without freeing
the resouce in some cases, and without notifying the caller.
Also, on FreeBSD with MALLOC_OPTIONS set i am seeing various panics
(duplicate freee etc.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
64a9c28c3b
do not ast_hangup() on a NULL channel.
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In the original code this would happen in the case of
o->forwards >= AST_MAX_FORWARDS
Likely an 1.2/1.4 isse as well - please someone have a look,
while I am hunting a few more similar panics now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
af51be05a6
Merged revisions 47850 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47850 | file | 2006-11-20 10:51:37 -0500 (Mon, 20 Nov 2006) | 2 lines
Use a separate variable in the channel structure to store the context that the channel was dialed from. (issue #8382 reported by jiddings)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47851 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Jason Parker
938c4bdc29
Merged revisions 47782 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47782 | qwell | 2006-11-16 17:19:46 -0600 (Thu, 16 Nov 2006) | 2 lines
Fix a couple of typos. Initially pointed out by mrobinson.
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19 years ago
Joshua Colp
a8ef79fa6f
Make local copy of arguments to parse. (issue #8362 reported by homesick)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
8b3fe4556c
move out another large block to a large function, and document
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some possibly missing parts in the privacy screening code.
Now that it is more streamlined it is easier to see differences
in handling the various cases.
Have not tested the code in depth.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47189 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
a55ca0d6ec
fix indentation of a block, and do minor simplifications at the end of
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another one.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
0f58d97707
complete previous commit.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
01ec7d63d9
move another block into a function.
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On passing, avoid two null-pointer string dereference
while printing messages (which are
sometimes not fatal in some platforms, but still wrong).
These two lines at least should be merged to 1.4 once i am
done with all the changes here.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47180 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
1385b6f047
move a large block into a separate function.
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Mark with XXX a possible bug in previous code which used
the wrong source in case of a forwarded call.
the function do_forward() needs to be split further, as the initial
part is replicated in another places (with some minor differences, most likely
forgotten when updating after the copy).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47179 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
ab4f699065
another small set of simplifications
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
acf5e87b0f
change HANDLE_CAUSE into a function.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
c3a6bbddb8
remove redundant checks
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47170 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
534da0ecfd
start integrating the simplifications proposed in bug 0005860,
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as usual a bit at a time to ease locating new bugs or fixes
worth merging into other branches.
In this commit, introduce a macro, S_REPLACE, that replaces
a string possibly freeing the previous value.
In one of these places (see the comment marked XXX) the previous
code might leak memory - if so, this ought to be merged in 1.4
The macro might be worth putting in one of the global headers
(e.g. include/asterisk/strings.h) as the construct is used
in a million places in the asterisk code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47160 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Steve Murphy
6d81c47801
These changes submitted by moy via bug 6992, to add a Dial 'End' event to asterisk. I include some changes to astman to cover other events that have been added.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
1e3c5bc5ba
Inherit the context and extension until the channel is answered
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Matt O'Gorman
ae8cc3e18b
bug #8076 check option_debug before printing to debug channel.
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patch provided in bugnote, with minor changes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
3e4a081e1c
Make callerid fields in Manager events more consistent. CallerIDNum for number and CallerIDName for name. (issue #7976 reported by suhler)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
1c764935f2
SS7 marked the start of an open season for trunk again but here's something minor - abstract early bridging into the technology so that we don't always assume they use RTP and try it.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Matt O'Gorman
d0a1a0033d
similar patch for verbose vs debug with minor changes
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bug 2617
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Steve Murphy
7c9b6e1d29
These small app documentation changes to app_dial and app_read will hopefully avert any more 7544 type bug reports\!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
0a27d8bfe5
merge new_loader_completion branch, including (at least):
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- restructured build tree and makefiles to eliminate recursion problems
- support for embedded modules
- support for static builds
- simpler cross-compilation support
- simpler module/loader interface (no exported symbols)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
774bba093b
Merged revisions 38928 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38928 | russell | 2006-08-05 02:37:59 -0400 (Sat, 05 Aug 2006) | 3 lines
make sure the priv-callerintros directory exists before trying to create a file
there (issue #7659 , patch by hads, with some modifications by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38929 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
6d0742fc16
merge Russell's 'hold_handling' branch, finally implementing music-on-hold handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Mark Spencer
4c90cf59b7
Support hold/unhold in Zap, update IAX2 parser to know about modern commands, forward hold/unhold in dial, add hold device state
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and implement holding in the SLA.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
BJ Weschke
2e8a142c85
Don't ast_request a channel structure twice when a call is being forwarded. ( #7362 - twlison / vechers confirming fix)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
a23af6559c
Merged revisions 33294 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r33294 | file | 2006-06-09 15:08:00 -0300 (Fri, 09 Jun 2006) | 2 lines
Handle hangup during recording of screened name (issue #7304 reported by kulldominique)
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19 years ago
Olle Johansson
9f5aa13142
Rename ast_rtp_early_media to ast_rtp_early_bridge to avoid confusion.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
472c1ca282
simplify autoconfig include mechanism (make tholo happy he can use lint again :-)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
7f3cc8b886
cleanups for commit from issue #5657... set a cause code for a rejected forward request, and actually set tmp->chan to NULL when we reject the forward request
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31049 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
BJ Weschke
871f08ec07
Add an option to app_dial, 'i', to instruct the application ignore any requests from peers to forward calls elsewhere. #5657 (johnlange w/some minor mods)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
91f9966be3
Merge branch for bug 6264 (Privacy option 2 returns dial-status ANSWER / option_priority_jumping not respected) (reported by jkoopmann and branch by murf)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30040 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
BJ Weschke
5235890be4
This is part 2/2 of the patches for #7090 . Adds one-step call parking to /trunk via builtin functions and 'k' 'K' application options added to app_dial. This also resolves #6340 .
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
d2da48b156
Inherit channel variables when call forwarding through chan_local (issue #7095 reported by raarts)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Russell Bryant
04ecb29d03
remove almost all of the checks of the result from ast_strdupa() or alloca().
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As it turns out, all of these checks were useless, because alloca will never
return NULL.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Mark Spencer
9953f4f40e
Make SIP early media work more efficiently without so many reinvites
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
BJ Weschke
bdf2a05aa5
Merged revisions 24567 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r24567 | bweschke | 2006-05-03 15:58:10 -0500 (Wed, 03 May 2006) | 3 lines
Correct application documentation to make users aware that certain options cannot be used in conjunction with others. #6666 (chotaire)
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19 years ago
Jim Dixon
a83297d85f
Added "Operator Services" connection mode for Zap channels, and the 'O' option
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in app_dial to support the use of this mode.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@22128 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
965ff42bed
more NULL "" equivalence in CID fields.
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Mark a potentially missing item in managerevent
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@21818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
b9f3e4e0f3
move a replicated block of code in the one place where it belongs.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@21596 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
bd01d66909
merge two nested 'if' which are really a single block.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@21595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago