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r327852 | mjordan | 2011-07-12 14:10:34 -0500 (Tue, 12 Jul 2011) | 12 lines
Added additional checks for mailbox / password beginning with '*' character
A bug existed such that if a user entered a password with '*', and the extension 'a' did not exist, an invalid mailbox would be created and the user authenticated. The code was changed to prevent this from occurring, and to prevent users from having mailboxes or passwords defined that begin with the '*' character.
(closes issue ASTERISK-17443)
Reported by: Kevin Scott Adams
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1316/
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When undergoing a shutdown and channels are kicked out of a bridge, a segfault
occurs because ConfBridge tries to play sounds on the bridge after the
underlying channels have been blown away due to the shutdown.
(closes ASTERISK-18040)
Review: https://reviewboard.asterisk.org/r/1283/
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r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) | 14 lines
Add the attribute "type" to each "<use>" for menuselect.
This matters only when autoconf fails to detect that weak linking is supported.
External optional dependencies will become optional in both cases, as they are
removed at compile time when not detected. However, runtime-optional modules
are made mandatory when weak linking is not found. This change affects only
the external optional dependencies; previously, they were incorrectly required
when weak linking support was not detected.
Patches:
20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003)
Tested by: iasgoscouk
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r325610 | rmudgett | 2011-06-29 13:05:15 -0500 (Wed, 29 Jun 2011) | 18 lines
Response to QueueRule manager command does not contain ActionID if it was specified.
* Add ActionID support as documented for the QueueRule AMI action.
* Remove documentation for ActionID with the Queues AMI action. The
output does not follow normal AMI response output and there is no place to
put an ActionID header.
(closes issue AST-602)
Reported by: Vlad Povorozniuc
Patches:
jira_ast_602_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Vlad Povorozniuc, rmudgett
Review: https://reviewboard.asterisk.org/r/1295/
JIRA SWP-3575
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* Added general option negative_penalty_invalid default off. when set
members are seen as invalid/logged out when there penalty is negative.
for realtime members when set remove from queue will set penalty to -1.
* Added queue option autopausedelay when autopause is enabled it will be
delayed for this number of seconds since last successful call if there
was no prior call the agent will be autopaused immediately.
* Added member option ignorebusy this when set and ringinuse is not
will allow per member control of multiple calls as ringinuse does for
the Queue.
- Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members
- QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty.
(closes issue ASTERISK-17421)
(closes issue ASTERISK-17391)
Reported by: irroot
Tested by: irroot, jrose
Review: https://reviewboard.asterisk.org/r/1119/
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When playing back a prompt to a channel, confbridge neglects to check for
hangup events causing lockup condititions for hangups that occur before
actually joining the conference. This change ensures that the user is removed
from the conference in the event of a premature hangup.
Review: https://reviewboard.asterisk.org/r/1277/
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Current behavior in ConfBridge menu definitions is that first definition takes
precedence, even in templated situations. This change allows inheritance and
overriding to work as expected so that the last definition takes precedence.
(closes ASTERISK-17986)
Review: https://reviewboard.asterisk.org/r/1267/
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This adds the playing_moh flag to the conference_bridge_user struct that
signifies when MOH should be playing so code doesn't have to guess whether
MOH is playing.
This change also adds the necessary checking to ensure that MOH continues
playing for a single user in a conference after the join sound is played when
configured to do so.
(closes ASTERISK-17988)
Review: https://reviewboard.asterisk.org/r/1263/
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Bridge and user profiles are not checked for existence before use. The lack
of a fully formed bridge profile can cause a segfault when sounds are accessed.
This change ensures that bridge and user profiles exist prior to usage
attempts.
Review: https://reviewboard.asterisk.org/r/1264/
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r322749 | rmudgett | 2011-06-09 11:31:53 -0500 (Thu, 09 Jun 2011) | 15 lines
Remove potential deadlock in call pickup race.
Deadlock is possible in ast_do_pickup() when holding the target channel
lock and trying to get the chan channel lock. Also, holding the target
lock when calling ast_channel_masquerade() is not a good idea because that
routine does deadlock avoidance.
* Removed the need to hold the target lock after marking the target with a
datastore and getting the connected line data off of the target channel.
* Moved can_pickup() to ast_can_pickup() in features.c. Now all the call
pickup methods use the same basic call pickup availability check.
Review: https://reviewboard.asterisk.org/r/1234/
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r322484 | rmudgett | 2011-06-08 15:46:55 -0500 (Wed, 08 Jun 2011) | 15 lines
Ring all queue with more than 255 agents will cause crash.
1. Create a ring-all queue with 500 permanent agents.
2. Call it.
3. Asterisk will crash.
The watchers array in app_queue.c has a hard limit of 255. Bounds
checking is not done on this array. No sane person should put 255 people
in a ring-all queue, but we should not crash anyway.
* Added bounds checking to the watchers array.
JIRA AST-464
JIRA SWP-2903
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r321330 | rmudgett | 2011-05-27 16:31:25 -0500 (Fri, 27 May 2011) | 8 lines
The app_privacy args have undocumented "options" position, interferes with "context" position.
* Add documention for unused "options" position to match existing code.
The trunk(v1.10) version will remove the unused options position.
(closes issue #19273)
Reported by: mdavenport
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r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) | 18 lines
The AMI Newstate event contains different information between v1.4 and v1.8.
The addition of connected line support in v1.8 changes the behavior of the
channel caller ID somewhat. The channel caller ID value no longer time
shares with the connected line ID on outgoing call legs. The timing of
some AMI events/responses output the connected line ID as caller ID.
These party ID's are now separate.
* The ConnectedLineNum and ConnectedLineName headers were added to many
AMI events/responses if the CallerIDNum/CallerIDName headers were also
present.
(closes issue #18252)
Reported by: gje
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1227/
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r320162 | jrose | 2011-05-20 13:12:21 -0500 (Fri, 20 May 2011) | 15 lines
Fixes an imapfolder related crash
imapfolders being set in the general section of voicemail would cause the inbox folder name to
change. Since sound file names are made based on the names of the folders, this would cause
the audio related to that folder name to change and if Asterisk attempted to play it, the
channel would instantly hang up when the audio file couldn't be found. This patch searches for
the name of the folder first to leave existing behavior in tact and if that fails, it uses
the normal inbox name to get the sound file instead.
(closes issue #16104)
Reported by: blkline
Review: https://reviewboard.asterisk.org/r/1215/
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r319997 | rmudgett | 2011-05-20 10:48:25 -0500 (Fri, 20 May 2011) | 25 lines
Crash when using directed pickup applications.
The directed pickup applications can cause a crash if the pickup was
successful because the dialplan keeps executing.
This patch does the following:
* Completes the channel masquerade on a successful pickup before the
application returns. The channel is now guaranteed a zombie and must not
continue executing the dialplan.
* Changes the return value of the directed pickup applications to return
zero if the pickup failed and nonzero(-1) if the pickup succeeded.
* Made some code optimizations that no longer require re-checking the
pickup channel to see if it is still available to pickup.
(closes issue #19310)
Reported by: remiq
Patches:
issue19310_v1.8_v2.patch uploaded by rmudgett (license 664)
Tested by: alecdavis, remiq, rmudgett
Review: https://reviewboard.asterisk.org/r/1221/
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r319529 | twilson | 2011-05-18 13:05:34 -0700 (Wed, 18 May 2011) | 24 lines
Merged revisions 319528 via svnmerge from
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r319528 | twilson | 2011-05-18 13:02:06 -0700 (Wed, 18 May 2011) | 17 lines
Merged revisions 319527 via svnmerge from
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r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011) | 10 lines
Fix app_dial ring groups
Revert part of r315643. We need to remove the datastore here as well.
The code in bridging code will catch anything that app_dial might miss.
(closes issue #19311)
Reported by: mspuhler
Patches:
issue_19311_no_answer.diff uploaded by elguero (license 37)
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r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines
Fix directed group pickup feature code *8 with pickupsounds enabled
Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues.
1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked.
2). dialplan applications for directed_pickups shouldn't beep.
3). feature code for directed pickup should beep on success/failure if configured.
Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite.
Moved app_directed:pickup_do() to features:ast_do_pickup().
Functions below, all now use the new ast_do_pickup()
app_directed_pickup.c:
pickup_by_channel()
pickup_by_exten()
pickup_by_mark()
pickup_by_part()
features.c:
ast_pickup_call()
(closes issue #18654)
Reported by: Docent
Patches:
ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585)
Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett
Review: https://reviewboard.asterisk.org/r/1185/
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r317969 | russell | 2011-05-06 16:49:01 -0500 (Fri, 06 May 2011) | 10 lines
Use the right variable to print the time in a debug message.
The original patch also increased some buffer sizes, but that was already
done in this version.
(closes issue #17034)
Reported by: sysreq
Patches:
asterisk-issue-17034.patch uploaded by sysreq (license 1009)
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r317584 | twilson | 2011-05-06 01:18:53 -0700 (Fri, 06 May 2011) | 20 lines
Merged revisions 317575 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r317575 | twilson | 2011-05-06 01:04:17 -0700 (Fri, 06 May 2011) | 13 lines
Merged revisions 317574 via svnmerge from
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r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011) | 6 lines
Re-fix queue round-robin
This part of the change for r315596 was incorrect. No bridge occurs
when doing a roundrobin dial and no one answers, so this code shouldn't
have been removed.
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r316831 | rmudgett | 2011-05-04 13:51:40 -0500 (Wed, 04 May 2011) | 9 lines
Wait for leader with Music On Hold allows crosstalk between participants.
Parenthesis in the wrong position. Regression from issue #14365 when
expanding conference flags to use 64 bits.
(closes issue #18418)
Reported by: MrHanMan
Tested by: rmudgett
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r316476 | seanbright | 2011-05-03 22:34:01 -0400 (Tue, 03 May 2011) | 17 lines
Merged revisions 316475 via svnmerge from
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r316475 | seanbright | 2011-05-03 22:23:01 -0400 (Tue, 03 May 2011) | 10 lines
Honor the C option to MeetMe when L is passed.
This fixes a case that r304773 and friends missed.
(closes issue #17317)
Reported by: var
Patches:
meetme-continue-on-l_16218.diff uploaded by var (license 1227)
Tested by: seanbright
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r315644 | twilson | 2011-04-26 14:39:01 -0700 (Tue, 26 Apr 2011) | 32 lines
Merged revisions 315643 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r315643 | twilson | 2011-04-26 14:27:44 -0700 (Tue, 26 Apr 2011) | 25 lines
Merged revisions 315596 via svnmerge from
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r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) | 18 lines
Allow transfer loops without allowing forwarding loops
We try to avoid the situation where two phones may be forwarded to each other
causing an infinite loop by storing each dialed interface in a channel
datastore and checking the list before dialing out. This works, but currently
breaks situations like A calls B, A transfers B to C, B transfers C to A, and A
transfers C to B. Since human interaction is happening here and not an
automated forwarding loop, it should be allowed.
This patch removes the dialed_interfaces datastore when a call is bridged (a
suggestion from the brilliant mmichelson). If a call is being bridged, it
should be safe to assume that we aren't stuck in a loop.
Since we are now handling this is the bridge code, the previous attempts at
handling it in app_dial and app_queue are removed.
Review: https://reviewboard.asterisk.org/r/1195/
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