Russell Bryant
dd920562ee
Clarify the documentation of the dialout and sendvoicemail options.
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(issue #9000 , caio1982 and serge-v)
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2007-03-06 23:00:57 +00:00
Olle Johansson
275abf4e08
Merged revisions 58052 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2 lines
Change error message to proper message
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2007-03-06 20:37:07 +00:00
Russell Bryant
905238839b
Return an error of transmit_response is called without a session.
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(issue #9002 )
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2007-03-06 18:01:20 +00:00
Joshua Colp
94c9dcda60
Since chan_iax2 does not support reception of DTMF with duration ensure that it is set to 0 on the frame. (issue #8521 reported by gdhgdh)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-05 19:19:07 +00:00
Joshua Colp
a5b62f15c3
Don't create a listen channel and record the conference unless the option is turned on. (issue #9204 reported by francesco_r)
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2007-03-05 18:39:28 +00:00
Joshua Colp
2feaf74cdd
Merged revisions 57869 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2 lines
Make create_dirpath use our standard for return values. -1 is failure, 0 is success. (issue #9205 reported by ballares)
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2007-03-05 17:52:03 +00:00
Steve Murphy
85b9c4f565
Merged revisions 57825 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1 line
Fixed a typo introduced via 9156 (either the gotos or their doc strings are wrong)
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2007-03-05 15:20:17 +00:00
Joshua Colp
127e1095f1
Don't allow a NULL pointer to reach ast_frdup. (issue #9155 reported by cmaj)
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2007-03-05 04:19:53 +00:00
Joshua Colp
a00f75604b
Don't reference a potentially NULL pointer. (issue #9199 reported by klolik)
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2007-03-05 03:35:03 +00:00
Joshua Colp
44dd29ff61
Preserve marker bit when P2P bridging. (issue #9198 reported by edgreenberg)
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2007-03-05 03:22:17 +00:00
Steve Murphy
b899145141
Updated the regression tests
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2007-03-03 15:31:12 +00:00
Tilghman Lesher
3687771760
Merged revisions 57648 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007) | 2 lines
Memory leak of a list, if call recording was abandoned
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2007-03-03 06:45:00 +00:00
Dwayne M. Hubbard
9a47a4712f
submitted patch for Georgian language, issue 9010, submitted by Alexander Shaduri
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2007-03-03 00:59:24 +00:00
Russell Bryant
3b6dc39807
add missing configuration template. Thanks to Lacy Moore on asterisk-users for pointing this out\!
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2007-03-03 00:02:29 +00:00
Russell Bryant
b59030489b
Update the check that is used to determine whether zaptel transcoder support
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is present. The interface has changed.
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2007-03-02 23:03:01 +00:00
Joshua Colp
53b9bc89c0
Merged revisions 57475 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2 lines
If a SIP message comes in and goes to a method handler that requires additional values that may not be present then send back an error.
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2007-03-02 17:06:52 +00:00
Steve Murphy
f8be0ce753
Merged revisions 57458 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1 line
further refinement in wording of goto documentation, as per 9156, goto not proceeding to next instruction
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2007-03-02 16:55:16 +00:00
Steve Murphy
9fab305234
I almost had comma escapes right, but 9184 points out the problem-- the escape is removed by pbx_config, and pbx_ael should also, before sending it down into the pbx engine. Also, you have to insert it back in, if you are generating extensions.conf code from the AEL.
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2007-03-02 05:21:36 +00:00
Russell Bryant
5a9cd3d235
Return the correct digit that interrupted the stream. This fixes exiting the
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Background application when using the m option.
(issue #9176 , mjagdis)
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2007-03-02 00:20:44 +00:00
Russell Bryant
31cf37519f
Merge changes from svn/asterisk/team/russell/sla_updates
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* Originally, I put in the documentation that only Zap interfaces would be
supported on the trunk side. However, after a discussion with Qwell, we came
up with a way to make IP trunks work as well, using some things already in
Asterisk. So, here it is, this now officially supports IP trunks.
* Update the SLA documentation to reflect how to setup IP trunks.
* Add a section in sla.txt that describes how to set up an SLA system with
voicemail.
* Simplify the way DTMF passthrough is handled in MeetMe.
* Fix a bug that exposed itself when using a Local channel on the trunk side
in SLA. The station's channel needs to be passed to the dial API when
dialing the trunk.
* Change a WARNING message to DEBUG in channel.h. This message is of no use
to users.
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2007-03-01 23:42:53 +00:00
Joshua Colp
219f4b4913
Merged revisions 57317 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar 2007) | 2 lines
Don't even attempt to optimize things when a proxy channel is involved. It will just explode in weird and unexplaineable ways. (issue #9175 reported by clegall_proformatique)
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2007-03-01 22:21:44 +00:00
TransNexus OSP Development
81b8f64769
1. Corrected a typo for www.etsi.org. Thank Patrick.
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2007-03-01 03:02:47 +00:00
Russell Bryant
65915e679a
minor tweaks to the sla docs
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2007-02-28 23:01:52 +00:00
Russell Bryant
447561d7a2
Merge more changes from svn/asterisk/team/russell/sla_updates
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* Add support for private hold. By setting "hold=private" for a trunk, only
the station that put the call on hold will be able to retrieve it from hold.
Also, by setting "hold=private" for a station, any call that station puts
on hold can only be retrieved by that station.
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2007-02-28 22:07:05 +00:00
Russell Bryant
315c9614cb
Minor formatting change
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2007-02-28 19:58:56 +00:00
Russell Bryant
9d3ff33b25
Merge changes from svn/asterisk/team/russell/sla_updates
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* Add support for the "barge=no" option for trunks. If this option is set,
then stations will not be able to join in on a call that is on progress
on this trunk.
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2007-02-28 19:56:20 +00:00
Steve Murphy
6bc3d77c05
Merged revisions 57118 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1 line
a small documentation update, to reflect reality in the goto doc strings, as per 9156, Goto does not proceed to next prio if jump fails
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2007-02-28 19:23:05 +00:00
Joshua Colp
43f841cadf
Merged revisions 57092 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb 2007) | 2 lines
Fix a few more issues with the agent logoff CLI command. (issue #9123 reported by arbrandes)
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2007-02-28 18:57:52 +00:00
Russell Bryant
9021d3c3b2
Merge current set of changes from svn/asterisk/team/russell/sla_updates
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* Add support for station ring delays. Ring delays can be set globally for a
station or for specific trunks on the station.
* Fix a few bugs in existing code.
* Restructure and Reorganize code to improve readability and maintainability.
* Improve formatting of the "sla show (trunks|stations)" CLI commands.
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2007-02-28 18:20:05 +00:00
Joshua Colp
53b014b81a
Picky compiler...
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2007-02-28 17:55:03 +00:00
Joshua Colp
d92e56ac89
Better handle timeouts when the individual speaks after everything has been played but before the timeout ends.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-28 17:45:50 +00:00
Steve Murphy
96622ddd26
I was surprised that I had not yet downgraded missing goto targets and macro call defs to a warning, in case they are in extensions.conf; I rectified this problem. Also, A goto in a macro to a target in a catch block was not being found; I fixed this too; the cause was that I needed to treat catch statements like an extension in the find_match code.
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2007-02-28 17:15:27 +00:00
Russell Bryant
1082f8d178
Fix voicemail email attachments. I missed the conversion of one of the line
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endings and there was an extra one where it should not have been.
(issue #9128 )
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2007-02-27 17:36:09 +00:00
Tilghman Lesher
91355b5e5f
Picky, picky... show deprecation warning in application help, too (reported via list)
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2007-02-26 22:01:23 +00:00
Russell Bryant
83ce3720bc
Restore the behavior of Asterisk 1.2 where if a device was not specified in
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alsa.conf, then we just use the system default, instead of creating our own
default of hw:0,0. (issue #9139 )
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2007-02-26 20:42:21 +00:00
Joshua Colp
31d02952de
Merged revisions 56850 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2 lines
Obey the clearglobalvars option in extensions reload (or dialplan reload depending on your version). (issue #9146 reported by ramonpeek)
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2007-02-26 20:07:18 +00:00
Russell Bryant
dd6b58edcd
Fix a crash in my last change to iax2_indicate(). (issue #9150 )
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2007-02-26 20:04:13 +00:00
Joshua Colp
0750349819
Update app_record documentation to use new CLI command, core show file formats. (issue #9151 reported by junky)
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2007-02-26 19:33:48 +00:00
Joshua Colp
51fc74da59
Use ast_strlen_zero to see if the language and/or context argument is not present for Background instead of just checking if it is NULL. (issue #9141 reported by mjagdis)
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2007-02-26 17:09:53 +00:00
Russell Bryant
268fc8a9cc
Do more complete locking of the chan_iax2_pvt struct in the indicate callback.
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(Problem brought up by Ben Smithurst on the asterisk-dev list)
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2007-02-26 16:51:18 +00:00
Joshua Colp
693cdc775e
Allow both of the show version files and core show file versions CLI commands to work. (issue #9135 reported by mvanbaak)
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2007-02-26 16:36:08 +00:00
Russell Bryant
11e4c71bc0
Move a comment to be in the correct struct.
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2007-02-26 01:04:40 +00:00
Russell Bryant
6c89f9e883
Blocked revisions 56729 via svnmerge
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r56729 | russell | 2007-02-25 18:34:31 -0600 (Sun, 25 Feb 2007) | 4 lines
Ensure that lock.h is included in utils.c with AST_API_MODULE defined so that
the implementations will be properly included when the AST_INLINE_API functions
are not going to be inlined. (issue #9124 , festr)
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2007-02-26 00:35:04 +00:00
Tilghman Lesher
185d7f5a3a
Merged revisions 56684 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007) | 3 lines
Issue 9130 - If prev is the last item on the channel list, then evaluating
additional conditions (e.g. name prefix) will cause a NULL dereference.
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2007-02-25 14:46:41 +00:00
Jason Parker
a9e4e0eefa
Make sure to set a speeddials parent on creation.
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Don't crash if hold is pressed when no call is active.
Don't return in places that we shouldn't..
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2007-02-24 02:02:53 +00:00
Kevin P. Fleming
3b191255bf
update to match zaptel 1.4 API change that was committed a few minutes ago
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2007-02-24 00:53:55 +00:00
Russell Bryant
2020191f67
Merged revisions 56504 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) | 8 lines
Fix up a couple more signal handlers to not do bad things that could cause
various undesirable results. The other day, I made Asterisk deadlock by
hitting Control-C because of a bad signal handler. Now, signal handlers
just set a flag and write to an alert pipe for the flag to be handled. Then,
there is another thread that is monitoring for these flags. If being run in
console mode, it is just the main thread. If Asterisk is in the background,
a thread is created to do it.
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2007-02-23 23:24:18 +00:00
Joshua Colp
40712e5eaf
Change log notice to debug. It is possible for a scheduled item to execute and be deleted at close to the same time and unavoidable. If this happens this message creeps up.
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2007-02-23 21:53:41 +00:00
Russell Bryant
76a643978e
Merged revisions 56406 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) | 4 lines
Don't destroy mutexes before unregistering all of the entry points from the core.
Also, fix a potential memory leak from not destroying the locks for all of the
possible call numbers (about 32k of them).
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2007-02-23 20:20:00 +00:00
Kevin P. Fleming
99b4865121
build special version strings for AADK/S800i builds
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2007-02-23 18:59:09 +00:00