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10.0.0
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10.10.0
10.10.0-digiumphones
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10.11.0
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10.11.0-digiumphones-rc1
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10.12.0
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10.12.1
10.12.1-digiumphones
10.12.2
10.12.2-digiumphones
10.12.3
10.12.3-digiumphones
10.12.4
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10.2.0
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10.5.0
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10.5.0-digiumphones-rc1
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10.5.1
10.5.1-digiumphones
10.5.2
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10.6.0
10.6.0-digiumphones
10.6.0-digiumphones-rc1
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10.7.0
10.7.0-digiumphones
10.7.0-digiumphones-rc1
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10.8.0
10.8.0-digiumphones
10.8.0-digiumphones-rc1
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10.9.0
10.9.0-digiumphones
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11.0.0
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11.5.1
11.6-cert11
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11.8.1
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12.0.0
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12.1.0
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12.6.0
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13.0.0
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13.7.1
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certified/1.8.11-cert1
certified/1.8.11-cert10
certified/1.8.11-cert2
certified/1.8.11-cert3-rc1
certified/1.8.11-cert3-rc2
certified/1.8.11-cert4
certified/1.8.11-cert5
certified/1.8.11-cert5-rc1
certified/1.8.11-cert5-rc2
certified/1.8.11-cert6
certified/1.8.11-cert7
certified/1.8.11-cert8
certified/1.8.11-cert9
certified/1.8.11-cert9-rc1
certified/1.8.15-cert1
certified/1.8.15-cert1-rc1
certified/1.8.15-cert1-rc2
certified/1.8.15-cert1-rc3
certified/1.8.15-cert2
certified/1.8.15-cert3
certified/1.8.15-cert4
certified/1.8.15-cert5
certified/1.8.15-cert6
certified/1.8.15-cert7
certified/1.8.28-cert1
certified/1.8.28-cert1-rc1
certified/1.8.28-cert2
certified/1.8.28-cert3
certified/1.8.28-cert4
certified/1.8.28-cert5
certified/1.8.6-cert1
certified/11.2-cert1
certified/11.2-cert1-rc1
certified/11.2-cert1-rc2
certified/11.2-cert2
certified/11.2-cert3
certified/11.6-cert1
certified/11.6-cert1-rc1
certified/11.6-cert1-rc2
certified/11.6-cert10
certified/11.6-cert11
certified/11.6-cert12
certified/11.6-cert2
certified/11.6-cert3
certified/11.6-cert4
certified/11.6-cert5
certified/11.6-cert6
certified/11.6-cert7
certified/11.6-cert8
certified/11.6-cert9
certified/13.1-cert1
certified/13.1-cert1-rc1
certified/13.1-cert1-rc2
certified/13.1-cert1-rc3
certified/13.1-cert2
certified/13.1-cert3
${ noResults }
21900 Commits (2ebbe7c31d8b9ca6b0d34b4c8ee6fc21e335596d)
Author | SHA1 | Message | Date |
---|---|---|---|
|
3efd0c7a5d |
Merged revisions 342276 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r342276 | kmoore | 2011-10-25 11:06:57 -0500 (Tue, 25 Oct 2011) | 18 lines Fix spool handling to allow call files to be hardlinked into place This fixes the inotify code to handle call files being hardlinked into the spool directory. The smsq utility does this, instead of rename(), to ensure that it cannot accidentally overwrite an existing spool file. A rename() might do that, but link() will definitely not. The inotify code had broken this, because it would wait for an IN_CLOSE_WRITE event on the file... which was never forthcoming, since it was never opened. Now we look for IN_OPEN events following the IN_CREATE event, and only wait for an IN_CLOSE_WRITE if the file was actually opened. Patch-by: dwmw2 (closes issue ASTERISK-18331) Review: https://reviewboard.asterisk.org/r/1391/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342277 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
2bdeae0faf |
Return NULL when no results returned for realtime_multientry
It was not documented what the return value should be when no entries were returned with the multientry realtime callback. This change forces consistent behavior even if the backends return an empty ast_config. Review: https://reviewboard.asterisk.org/r/1521/ ........ Merged revisions 342223 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342224 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
06f43f5c04 |
Fix ao2obj.h comment typos and add missing link/unlink nolock debug defines.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342183 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
97e6bbf7fd |
Outbound SIP OPTIONS messages will now include fromuser of related peer.
This behavior matches up more closely with the way invite/register/etc are handled. This patch also modifies some adjacent code for code style compliance. Pretty minor. (closes issue ASTERISK-17616) Reported by: Jeremy Kister Patches: chan_sip.c-options-fromuser-fix-v1.patch uploaded by Jeremy Kister (license #6232) ........ Merged revisions 342061 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342062 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
55a92b7499 |
queues container needs locking when using the OBJ_NOLOCK flag
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@342017 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
05f7d0c095 |
Remove some ref leaks and a return without unlock.
There some resource leaks introduced in asterisk 10 make sure that locks are not held on return and we release ref's held. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341972 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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b0120e5e43 |
Revert Janitor patch 341920 For now
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341922 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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ec07b6448c |
Whitespace Fixups / Add Braces
This janitorial patch is related to work on RB1538 ........ Merged revisions 341906 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341920 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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dff70e96d6 |
only process args that exist
ASTERISK-18395 ........ Merged revisions 341809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341810 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c364c27f5c |
don't limit the length of app and function arguments
ASTERISK-18395 ........ Merged revisions 341806 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341807 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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ce19768cec |
Fix AGI exec Park to honor the Park application parameters.
The fix for ASTERISK-12715 and ASTERISK-12685 added a check for the Park application because the channel needed to be masqueraded to prevent a crash. Since the Park application now always masquerades the channel into the parking lot, the special check is no longer needed. The fix also resulted in AGI exec Park attempting to double park the call and not honor the Park application parameters. * Removed no longer necessary call to ast_masq_park_call() by AGI exec for the Park application. (Reverts -r146923) * Fix Park application to only return 0 or -1. The AGI exec Park was causing broken pipe error messages because the Park application returned 1 on successful park. (closes issue ASTERISK-18737) ........ Merged revisions 341717 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341718 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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467d40c0b6 |
Fixed typo from previous commit
........ Merged revisions 341704 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341707 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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986c3e6c64 |
Updated documentation for the optional CID parameter with CALLERID
........ Merged revisions 341664 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341665 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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feb1211d78 |
add documentation for check_state_unknown in configs/queues.conf.sample
app_queue allows calls to members in a "Unknown" state to be treated as available setting check_state_unknown = yes will cause app_queue to query the channel driver to better determine the state this only applies to queues with ringinuse or ignorebusy set appropriately. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341599 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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3aa9147fc2 |
Add option to check state when state is unknown
r341486 reverts r325483 this is a rework of the patch. optimize to minimize load. add option check_state_unknown to control whether a member with unknown device state is checked there is a small % chance that calls will be sent to the member when they on a call. app_queue will see a device with unknown state as available and does not try verify the state without this option enabled. Review: https://reviewboard.asterisk.org/r/1535/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341580 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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b761c1eef2 |
Clean up ast_check_digits
The code was originally copied from the is_int() function in the AEL code. wdoekes pointed out that the function should take a const char* and that their was an unneeded variable. This is now fixed. ........ Merged revisions 341529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341530 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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69d2d46275 |
Fix a performance regression introduced in r325483.
The regression was caused by a call to ast_parse_device_state() in app_queue's ring_entry() function. The ast_parse_device_state() function eventually calls ast_channel_get_full() with a channel name prefix which causes it to walk the channel list causing massive lock contention and slow downs. This patch fixes the regression by removing the call to ast_parase_device_state() which should be unnecessary. Queue member device state should be maintained by device state events. Some users have seen instances where busy agents were called when they shouldn't have, which is the reason the call to ast_parse_device_state() was added. That change appears to have resolved that issue but also causes this performance regression. There may still be issues with queue member status, and if so, alternative methods should be investigated to resolve them. AST-695 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341486 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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f62f95c1b4 |
Outgoing calls with Google Voice
Google has recently make some changes (again) to their protocol. Rather then patching asterisk to flip between the two different methods, we now allow both. Lets hope this keeps Google Voice happy for a while. (closes issue ASTERISK-18714) Reported by: Iordan Iordanov Patches: chan_gtalk.patch uploaded by Iordan Iordanov (licenses 6311) ........ Merged revisions 341435 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341436 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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26f196468f |
Don't use is_int() since it doesn't link well on all platforms
Just create an normal API function in strings.h that does the same thing just to be safe. ASTERISK-17146 ........ Merged revisions 341379 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341380 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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a4cc7f518e |
Don't sent in-dialog requests like UPDATE when Asterisk has not yet received a Contact URI from a UAS
........ Merged revisions 341366 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341377 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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aa0c65619d |
Don't resolve numeric hosts or contact unresolved hosts
If a SIP dial string contains a numeric hostname that is not a peer name, don't try to resolve it as it is unlikely that someone really means Dial(SIP/0.0.4.26) when Dial(SIP/1050) is called. Also, make sure that create_addr returns -1 if an address isn't resolved so that we don't attempt to send SIP requests to an address that doesn't resolve. (closes issue ASTERISK-17146, ASTERISK-17716) Review: https://reviewboard.asterisk.org/r/1532/ ........ Merged revisions 341314 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341315 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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21edfbb5fd |
Merged revisions 341312 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r341312 | may | 2011-10-19 03:20:53 +0400 (Wed, 19 Oct 2011) | 3 lines fix issue on channel numbering (calls could have same channel number on heavy loaded system) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341313 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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fa58ec2c74 |
More parking issues.
* Fix potential deadlocks in SIP and IAX blind transfer to parking. * Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the parkext_exclusive option with transfers (Park(,,,,,exclusive_lot) parameter). Created ast_park_call_exten() and ast_masq_park_call_exten() to maintian API compatibility. * Made masq_park_call() handle a failed ast_channel_masquerade() setup. * Reduced excessive struct parkeduser.peername[] size. ........ Merged revisions 341254 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341255 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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fe6dd5b23a |
Initialize variables before calling parse_uri
If parse_uri was called with an empty URI, some pointers would be modified and an invalid read could result. This patch avoids calling parse_uri with an empty contact uri when parsing REGISTER requests. AST-2011-012 (closes issue ASTERISK-18668) ........ Merged revisions 341189 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341190 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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bbc5c7826e |
Remove an unused include of md5.h
Unused include of asterisk/md5.h in pbx_realtime.c . A commit needed to test the commit message. Merged-From: http://svn.asterisk.org/svn/asterisk/branches/1.8@341074 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341148 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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a4ce115a56 |
Set 'core' support level for test_format_api.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341146 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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824c5dbfa2 |
Multiple revisions 341108,341112
........ r341108 | pabelanger | 2011-10-17 12:22:19 -0400 (Mon, 17 Oct 2011) | 2 lines Voicemail compiler flags are 'core' support ........ r341112 | pabelanger | 2011-10-17 12:23:33 -0400 (Mon, 17 Oct 2011) | 2 lines Fix previous commit ........ Merged revisions 341108,341112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341122 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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db7a2781da |
Add information about limitations of new codec support in channel drivers.
(issue ASTERISK-18680) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341094 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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f99e9b5bad |
Don't try to remove peers without IPs from peers_by_ip
(closes issue ASTERISK-18696) ........ Merged revisions 341088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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7a1f2ca356 |
Change the internal name of the menuselect options that are used to control
whether modules are embedded or not; using just the bare category name led to accidentally enabling these options when users used the wrong "--enable" operation on the menuselect command line. Now the internal option names are prefixed with "EMBED_", so they won't be the same as the name of the category containing the modules they control the embedding of. ........ Merged revisions 341022 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@341023 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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91a65d8b3f |
Merged revisions 340970 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) | 8 lines Quiet RTCP Receiver Reports during fax transmission RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions. The ability to disable RTCP streams in res_rtp_asterisk was missing, so this code was added to support the bug fix. (closes issue ASTERISK-18400) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340971 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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ad8129697a |
Some additional module documentation changes for 10 for the menuselect change.
(issue ASTERISK-18268) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340931 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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cb6e65cdf3 |
Avoid unnecessary WARNING message
Add AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid displaying a WARNING message. (closes issue ASTERISK-18610) Patch by: Kristijan_Vrban ........ Merged revisions 340878 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340879 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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d3831204a9 |
Fixes some support level info so that it can be read by menuselect.
(issue ASTERISK-18268) Review: https://reviewboard.asterisk.org/r/1525/ ........ Merged revisions 340863 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340868 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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3d8a367788 |
Fix DTMF blind transfer continuing to execute dialplan after transfer.
Party A calls Party B. Party A DTMF blind transfers Party B to Party C. Party A channel continues to execute dialplan. * Fixed the return value of builtin_blindtransfer() to return the correct value after a transfer so the dialplan will not keep executing. * Removed unnecessary connected line update that did not really do anything. * Made access to GOTO_ON_BLINDXFR thread safe in check_goto_on_transfer(). * Fixed leak of xferchan for failure cases in check_goto_on_transfer(). * Updated debug messages in builtin_blindtransfer() and check_goto_on_transfer(). (closes issue ASTERISK-18275) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 340809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340810 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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e372d326ca |
Only send MWI Notify on register if the registration is successful.
lastmsgssent was removed from chan_sip and the old behavior of sending a mwi notify on register [except when subscribemwi is set] was restored but this must only happen when registration succeeds. leaking information for unsuccessful registrations is not secure. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340770 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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cdae7c1714 |
Merged revisions 340717 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340717 | schmidts | 2011-10-13 06:58:00 +0000 (Thu, 13 Oct 2011) | 3 lines storing the route-set also on a 181 response not only on 180,182 or 183. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340718 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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e00bacf5aa |
Initialize ast_sockaddr before calling ast_sockaddr_resolve
Avoid possible jump based on unitialized value ........ Merged revisions 340715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340716 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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4e1084a392 |
Don't skip the query field on a realtime multi query
There is no documented reason to not add the query field to the varlist returned by a realtime multi query, despite the config category being set to its value. Of course, there is no documentation that the category should be set to the value either. There is lots of no documentation when it comes to realtime. But, other engines do not skip this field so I am forcing this backend to follow the convention, because not doing so is very silly. ........ Merged revisions 340662 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340663 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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925bcb92b4 |
Merged revisions 340534 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340534 | twilson | 2011-10-12 13:19:36 -0700 (Wed, 12 Oct 2011) | 9 lines Update SIP realtime fullcontact regardless of caching We should update the fullcontact field in the realtime table whether or not rtcachefriends is set. There is no reason to treat a non-cached realtime entity differently than a cached in this regard. (closes issue ASTERISK-18446) Reported by: wdoekes ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340578 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
0482a757c3 |
Merged revisions 340576 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340576 | schmidts | 2011-10-12 20:30:37 +0000 (Mit, 12 Okt 2011) | 3 lines Store route-set from provisional SIP responses so early-dialog requests can be routed properly ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340577 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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dc8a317f87 |
Initialize the PRI channel alarms properly on startup.
The PRI channel alarms were initialized with an inverted sense. (closes issue ASTERISK-18710) Reported by: Tzafrir Cohen ........ Merged revisions 340522 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340523 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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ec1778c05f |
Update MeetMe p and X option documentation when interacting with the s option.
ASTERISK-12175 changed the p and X options to not interfere with the s option when they are used together. It makes more sense for the s option to have priority for the DTMF '*' key since it cannot change its activation code. Otherwise, you could not use option s with the p or X options. JIRA AST-671 ........ Merged revisions 340470 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340471 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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287c621613 |
Fix verbose messages when IPv6 logic was added
(closes issue ASTERISK-18612) Reported by: Tim Osman ........ Merged revisions 340418 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340419 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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dbd7f17342 |
Add protection for SS7 channel allocation and better glare handling.
* Added a CLI "ss7 show channels" command that might prove useful for future debugging. * Made the incoming SS7 channel event check and gripe message uniform. * Made sure that the DNID string for an incoming call is always initialized. (issue ASTERISK-17966) Reported by: Kenneth Van Velthoven Patches: jira_asterisk_17966_v1.8_glare.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 340365 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340366 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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baa5ff86e5 |
Fix some potential deadlocks pointed out by helgrind.
* Fixed deadlock potential calling dialog_unlink_all() in __sip_autodestruct(). Found by helgrind. * Fixed deadlock potential in handle_request_invite() after calling sip_new(). Found by helgrind. * The sip_new() function now returns with the created channel already locked. * Removed the dead code that starts a PBX in in sip_new(). No sip_new() callers caused that code to be executed and it was a bad thing to do anyway. * Removed unused parameters and return value from dialog_unlink_all(). * Made dialog_unlink_all() and __sip_autodestruct() safely obtain the owner and private channel locks without a deadlock avoidance loop. ........ Merged revisions 340284 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340310 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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4348fc2cc9 |
Convert registered AMI actions to ao2 objects.
* Fixed race between calling an AMI action callback and unregistering that action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change. * Fixed potential memory leak if an AMI action failed to get registered because is already was registered. Part of the ao2 conversion. * Fixed AMI ListCommands action not walking the actions list with a lock held. * Fix usage of ast_strdupa() and alloca() in loops. Excess stack usage. * Fix AMI Originate action Variable header requiring a space after the header colon. Reported by Yaroslav Panych on the asterisk-dev list. * Increased the number of listed variables allowed per AMI Originate action Variable header to 64. * Fixed AMI GetConfigJSON action output format. * Fixed usage of res contents outside of scope in append_channel_vars(). * Fixed inconsistency of config file channelvars option. The values no longer accumulate with every channelvars option in the config file. Only the last value is kept to be consistent with the CLI "manager show settings" command. (closes issue ASTERISK-18479) Reported by: Jaco Kroon ........ Merged revisions 340279 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340281 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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3275275a07 |
Update SHA1 code to RFC 6234
RFC 6234 is an update to RFC 3174 from which the code was originally taken. It has a slightly better code, and a better phrased license (simple 3-clause BSD). * main/sha1.c is sha1.c from RFC 6234 with formatting changes only. * include/asterisk/sha1.h merges sha.h and sha-private.h from RFC 6234. * Removed unused include of asterisk/sha1.h from main/channels.c Review: https://reviewboard.asterisk.org/r/1503/ Merge-From: http://svn.asterisk.org/svn/asterisk/branches/1.8@340263 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340280 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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1e1100cdb4 |
On astdb conversion, also warn about permissions requirements
The user running Asterisk must have permission to the directory the Asterisk database resides in since SQLite 3 needs to be able to create a journal file. (closes issue ASTERISK-18174) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340222 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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0d1ff9db85 |
Add a missing file for the astdb2bdb conversion utility
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@340220 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |