Commit Graph

277 Commits

Author SHA1 Message Date
Mark Michelson e4cc4ef183 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 20:03:48 +00:00
Mark Michelson 402820c9e3 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 18:16:15 +00:00
Tilghman Lesher 3a215d6a50 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 12:38:53 +00:00
Tilghman Lesher badad4a542 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 19:29:23 +00:00
Joshua Colp f07dfca43a 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 14:21:59 +00:00
Joshua Colp 1317920054 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 17:59:14 +00:00
Steve Murphy dfee354cfa 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 22:06:27 +00:00
Russell Bryant 78989c792f 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 16:47:29 +00:00
Joshua Colp 03eb572457 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 19:17:08 +00:00
Joshua Colp 88e5a094b6 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 01:03:14 +00:00
Joshua Colp ec6e4f6887 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 03:55:30 +00:00
Joshua Colp 3a1ceb4ff8 Need to check macro extension as well as macro context for directed pickup.
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2007-02-16 18:51:15 +00:00
Joshua Colp 7722d96bea 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 17:02:35 +00:00
Joshua Colp bddfe6fea7 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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2007-02-15 16:19:39 +00:00
Joshua Colp dadf652e26 Forward begin DTMF frames as well as end. (issue #9068 reported by mhardeman)
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2007-02-14 21:07:23 +00:00
Joshua Colp 1ba2aa702d Temporarily change musicclass on channel to one specified in Dial so that the 'm' option functions properly. (issue #8969 reported by christianbee)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-09 19:33:31 +00:00
Russell Bryant 8f04a45ee9 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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2007-02-03 20:44:20 +00:00
Joshua Colp 2c4359d9b3 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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2007-01-10 04:55:13 +00:00
Joshua Colp 8d7bdfc607 Merged revisions 48584 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r48584 | file | 2006-12-19 16:10:26 -0500 (Tue, 19 Dec 2006) | 2 lines

Free localuser structure when we fail to dial (issue #8612 reported by rizzo)

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2006-12-19 21:12:07 +00:00
Kevin P. Fleming a2f6499c0a 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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2006-12-01 23:37:28 +00:00
Joshua Colp 335630b10c 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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2006-11-20 15:51:37 +00:00
Jason Parker 67546a068c Fix a couple of typos. Initially pointed out by mrobinson.
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2006-11-16 23:19:46 +00:00
Joshua Colp 552ca7e926 Backport of new directed pickup (BE-85).
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2006-10-16 17:24:35 +00:00
Matt O'Gorman d0a1a0033d similar patch for verbose vs debug with minor changes
bug 2617


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2006-09-19 16:23:45 +00:00
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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2006-09-06 17:14:50 +00:00
Kevin P. Fleming 0a27d8bfe5 merge new_loader_completion branch, including (at least):
- 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)



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2006-08-21 02:11:39 +00:00
Russell Bryant 774bba093b Merged revisions 38928 via svnmerge from
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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2006-08-05 06:39:43 +00:00
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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2006-07-19 20:44:39 +00:00
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
and implement holding in the SLA.


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2006-07-08 02:24:07 +00:00
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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2006-06-16 12:18:18 +00:00
Joshua Colp a23af6559c Merged revisions 33294 via svnmerge from
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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2006-06-09 18:12:46 +00:00
Olle Johansson 9f5aa13142 Rename ast_rtp_early_media to ast_rtp_early_bridge to avoid confusion.
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2006-06-09 09:47:44 +00:00
Kevin P. Fleming 472c1ca282 simplify autoconfig include mechanism (make tholo happy he can use lint again :-)
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2006-06-07 18:54:56 +00:00
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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2006-05-31 16:11:55 +00:00
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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2006-05-31 15:52:32 +00:00
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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2006-05-24 20:00:10 +00:00
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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2006-05-22 16:43:43 +00:00
Joshua Colp d2da48b156 Inherit channel variables when call forwarding through chan_local (issue #7095 reported by raarts)
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2006-05-16 23:39:29 +00:00
Russell Bryant 04ecb29d03 remove almost all of the checks of the result from ast_strdupa() or alloca().
As it turns out, all of these checks were useless, because alloca will never
return NULL.


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2006-05-10 13:22:15 +00:00
Mark Spencer 9953f4f40e Make SIP early media work more efficiently without so many reinvites
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2006-05-09 11:44:50 +00:00
BJ Weschke bdf2a05aa5 Merged revisions 24567 via svnmerge from
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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2006-05-03 21:11:23 +00:00
Jim Dixon a83297d85f Added "Operator Services" connection mode for Zap channels, and the 'O' option
in app_dial to support the use of this mode.


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2006-04-22 11:30:06 +00:00
Luigi Rizzo 965ff42bed more NULL "" equivalence in CID fields.
Mark a potentially missing item in managerevent



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2006-04-21 10:41:13 +00:00
Luigi Rizzo b9f3e4e0f3 move a replicated block of code in the one place where it belongs.
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2006-04-19 18:15:20 +00:00
Luigi Rizzo bd01d66909 merge two nested 'if' which are really a single block.
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2006-04-19 18:07:19 +00:00
Luigi Rizzo 9476cb356e fix indentation of a large block
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@21572 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-04-19 18:00:32 +00:00
Luigi Rizzo 501c9e181c start sorting out the duplicated code in the privacy handler
for future removal



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2006-04-19 17:58:07 +00:00
Luigi Rizzo b3343aecdf merge two nested 'if' which are really a single block.
(indentation still to be fixed)



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2006-04-19 17:29:15 +00:00
Luigi Rizzo 73f2d344fb more localization and variable removal
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2006-04-19 16:54:04 +00:00
Luigi Rizzo c9f669e56d more localization of variables
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2006-04-19 16:36:15 +00:00