Joshua Colp
76f8e0c489
Fix spelling of unavailable in voicemail documentation. (issue #9248 reported by tensai)
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2007-03-09 21:08:19 +00:00
Joshua Colp
b9124234e9
Merged revisions 58579 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2 lines
If we are unable to lookup the host in a c line we have to abort, otherwise the previous data is gone and we will (potentially) have no data when all is said and done.
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2007-03-09 20:49:47 +00:00
Russell Bryant
90aa4ab658
Hang up the channel that put the call on hold in the event processing thread to
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avoid a race condition. Also, if the station originated the call that it is
putting on hold, don't hang up the trunk if it was the only station on the call
and it is hanging up due to hold and not a normal hangup.
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2007-03-08 22:15:15 +00:00
Russell Bryant
2f54661cf8
Add a missing break statement so that handling the above event does not
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incorrectly destroy the channel. (issue #9242 , andrew)
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2007-03-08 22:06:54 +00:00
Tilghman Lesher
ec21f50132
Fix segfault (Issue 9236)
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2007-03-08 21:33:03 +00:00
Russell Bryant
ed9b9c6791
Refactor hold handling a bit so that it does not require keeping the call up
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when a call is put on hold.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-08 20:54:56 +00:00
Joshua Colp
2b3c0dc1e4
Make early SDP seeding even smarter! We have to check codecs in the make_compatible function too. (issue #9221 reported by marcelbarbulescu)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-08 18:01:00 +00:00
Joshua Colp
add0cad065
Merged revisions 58388 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2 lines
Only print out debug message if the definition that makes the variables shows up was actually defined. (issue #9233 reported by serginuez)
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2007-03-08 16:07:10 +00:00
Kevin P. Fleming
690da7f074
this change was not needed; fclose() handles closing the file descriptor already
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2007-03-08 13:23:46 +00:00
Kevin P. Fleming
37af77e79f
fix a compiler warning, and overwriting 'res' value
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2007-03-08 13:17:42 +00:00
Kevin P. Fleming
8a2e3d0b8b
fix two cases where HTTP session file descriptors would not be closed
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-08 13:17:17 +00:00
Russell Bryant
5603ca69eb
If we receive ZT_EVENT_REMOVED, destroy the specified channel.
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(issue #7256 , tzafrir)
Also, update the configure script to make sure that we don't try to build
chan_zap if the installed version of zaptel does not include ZT_EVENT_REMOVED.
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2007-03-08 01:01:46 +00:00
Russell Bryant
f34c30860e
(This bug was reported to me by Kinsey Moore)
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Merged revisions 58242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) | 7 lines
Fix a problem where the Asterisk channel name could be that of the wrong IAX2
user for a call. This is because the first step of choosing this name is to
look for an IAX2 peer that happens to have the same IP/port number that this
call is coming from and assuming that is it. However, this is not always
correct. So, I have made it change this name after authentication happens
since at that point, we have an exact match.
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2007-03-07 18:19:19 +00:00
Joshua Colp
240355d78b
Ensure we have (or should have) at least one matching codec before attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu)
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2007-03-07 17:52:58 +00:00
Russell Bryant
b393ff7feb
Blocked revisions 58167 via svnmerge
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r58167 | russell | 2007-03-06 18:27:04 -0600 (Tue, 06 Mar 2007) | 2 lines
Fix a misplaced block of code in the 1.2 version of the patch to fix issue #8977
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2007-03-07 00:27:59 +00:00
Russell Bryant
97fe798e7e
Merged revisions 58164 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) | 4 lines
If the channels acquired using the manager Redirect action are not up, then
don't attempt to do anything with them. It could lead to weird behavior,
including crashes. (issue #8977 )
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2007-03-07 00:25:19 +00:00
Steve Murphy
81ef9a4f18
Merged revisions 58115 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1 line
Fix for 9220: Eyebeam cannot renew subscriptions for presence info. Reason: re-SUBSCRIBE requests don't include Accept headers, which the rfc says are optional (to put it tersely), (it uses MAY), and luckily, the sip_pvt struct has the format info stored, so we simply leave it if the format is set, and the accept header null.
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2007-03-06 23:10:14 +00:00
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57872 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57473 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57426 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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