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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-06 21:52:30 +00:00
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-01 15:39:54 +00:00
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
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");
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-26 15:49:18 +00:00
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
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
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
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
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
ce8f95d750
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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
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
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
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.
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2007-07-09 08:27:37 +00:00
8b93f50dfc
Merged revisions 73053 via svnmerge from
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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
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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
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
2462d5ab4f
Cleaning up a small disaster I created earlier
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2007-06-20 23:26:07 +00:00
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
ce2c52d519
Merged revisions 70445 via svnmerge from
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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
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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
704c756c4a
Merge work to make U(...) option work for Dial
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2007-06-20 17:35:08 +00:00
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
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
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
2492bf26fb
Merged revisions 68071 via svnmerge from
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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
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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
9d05ff8ed5
Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
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2007-06-06 21:20:11 +00:00
bae04fd90e
Merged revisions 67066 via svnmerge from
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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
4572edae31
Merged revisions 65200 via svnmerge from
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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
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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
83d0b0417c
Merged revisions 64756 via svnmerge from
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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
e2871220e2
Merged revisions 61656 via svnmerge from
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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
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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
be874b92d3
Remove unused instances of unnamed enums.
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2007-04-09 22:49:32 +00:00
840f1e61e0
Merged revisions 60798 via svnmerge from
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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
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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
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
fb9e48ca9e
Merged revisions 55154 via svnmerge from
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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
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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
8e92c73a1d
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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
821c941976
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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
a11b56a8e5
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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
68a66656e6
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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
32cd307d6f
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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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2007-02-09 19:39:26 +00:00
ce321f87e9
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r53136 | russell | 2007-02-03 14:44:20 -0600 (Sat, 03 Feb 2007) | 12 lines
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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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2007-02-03 20:46:36 +00:00
a378353ecc
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r50298 | file | 2007-01-09 23:55:13 -0500 (Tue, 09 Jan 2007) | 10 lines
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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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2007-01-10 04:56:48 +00:00
83f52ed5e2
better name for struct dial_localuser.
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2006-12-19 16:36:45 +00:00
a755ec928a
introduce a temporary variable for tmp->chan to shorten expressions.
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2006-12-19 09:58:40 +00:00
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.
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2006-12-19 09:33:57 +00:00
9d4531d636
move a large block related to privacy handling to a separate function.
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2006-12-19 09:15:23 +00:00
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r48193 | kpfleming | 2006-12-01 17:37:28 -0600 (Fri, 01 Dec 2006) | 10 lines
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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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2006-12-01 23:39:59 +00:00
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
2006-11-21 11:53:06 +00:00
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
2006-11-21 11:07:30 +00:00
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
2006-11-20 15:55:58 +00:00
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-16 23:20:52 +00:00
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
2006-11-14 20:09:10 +00:00
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
2006-11-04 11:00:49 +00:00
Joshua Colp
Russell Bryant
Steve Murphy
Tilghman Lesher
Mark Michelson
Jason Parker
Olle Johansson
Luigi Rizzo
Kevin P. Fleming