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r365395 | qwell | 2012-05-04 16:17:08 -0500 (Fri, 04 May 2012) | 7 lines
Add support for folders in MixMonitor 'm' option. Backport manager actions.
The manager actions are needed, so MixMonitor can be executed on existing
channels.
(issue DPMA-68)
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r364761 | qwell | 2012-05-01 12:25:14 -0500 (Tue, 01 May 2012) | 6 lines
Remove folder_dir from voicemail snapshots API.
It was both unused (except in tests, where it was fudged) and unnecessary.
(closes issue AST-842)
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r367161 | mmichelson | 2012-05-21 14:05:52 -0500 (Mon, 21 May 2012) | 21 lines
Add "send to voicemail" Digium phone functionality to Asterisk.
This change accommodates two methods by which calls can be directed to
a user's voicemail.
* Incoming calls can be redirected to any user's voicemail.
* Established calls can be blind transferred to any user's voicemail.
Digium phones indicate the desire to direct a call to voicemail by using
a Diversion header with a reason parameter of "send_to_vm".
This patch adds the "send_to_vm" reason as a valid redirecting reason. In
addition, chan_sip.c has been modified to update redirecting information
on the transferred channel by reading a Diversion header on a REFER request.
(closes issue AST-871)
Reported by Malcolm Davenport
Review: https://reviewboard.asterisk.org/r/1925
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r368790 | mjordan | 2012-06-12 08:44:36 -0500 (Tue, 12 Jun 2012) | 18 lines
Fix deadlock in SIP transfers that involve a REFER request
In r367163, "send to voicemail" functionality was added to the SIP channel
driver. This required updating the party redirecting information for the
channel based on the headers provided in the REFER request. When the
redirecting party information is updated on the channel, a call to
ast_indicate_data occurs. Because handle_request_refer still had the sip_pvt
locked, a deadlock could occur between the pbx_thread and the do_monitor thread
servicing the REFER request.
This patch preserves the proper locking order between the channel and the
sip_pvt by ensuring that the sip_pvt is unlocked prior to updating the party
redirecting information on the channel.
(closes issue AST-903)
Reported by: Matt Jordan
patches:
jira_ast_903_trunk.patch by rmudgett (license 5621)
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r368962 | qwell | 2012-06-14 13:38:48 -0500 (Thu, 14 Jun 2012) | 11 lines
Remove global symbol requirement from app_voicemail.
This uses the existing "function installation" stuff that already existed for
other functions, like getting message counts.
(closes issue AST-807)
(issue AST-901)
(issue AST-908)
Review: https://reviewboard.asterisk.org/r/1965/
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r368964 | qwell | 2012-06-14 14:03:24 -0500 (Thu, 14 Jun 2012) | 8 lines
These functions that were moved need to be static.
Also wrap test functions in a #ifdef.
(issue AST-807)
(issue AST-901)
(issue AST-908)
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r368998 | qwell | 2012-06-15 10:31:43 -0500 (Fri, 15 Jun 2012) | 6 lines
Remove some symbol exports that got missed in the removal of global symbols.
(issue AST-807)
(issue AST-901)
(issue AST-908)
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r369024 | qwell | 2012-06-15 11:29:40 -0500 (Fri, 15 Jun 2012) | 2 lines
Fix voicemail API tests by using the correct argument order for create/destroy.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/certified/branches/1.8.11@369839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r354890, a memory leak in app_voicemail was fixed by properly disposing of
the allocated heard/deleted pointers. However, there are situations,
particularly when no messages are found in a folder, where these pointers are
not allocated and not NULL. In that case, an invalid free would be attempted,
which could crash app_voicemail. As there are a number of code paths where
this could occur, this patch uses the number of messages detected in the folder
before it attempts to free the pointers. This resolves the crash detected in
the Asterisk Test Suite's check_voicemail_nominal test.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356797 65c4cc65-6c06-0410-ace0-fbb531ad65f3
open_mailbox() was changed quite a long time ago to allocate this memory.
close_mailbox() should have been changed to be responsible for freeing it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The voicemail [general] zonetag and locale variables weren't loaded
until after the mailboxes were initialized. This caused the settings to
be unset for those mailboxes until a reload was performed.
(closes issue ASTERISK-18838)
Review: https://reviewboard.asterisk.org/r/1570
Reviewed by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Before, using the U flag in Voicemail with multiple recipients would put urgent messages
in the INBOX folder for all users past the first thanks to a bug with the message
copying function. This would also cause messages to fail to be sent if the INBOX
directory hadn't been created for that mailbox yet.
(closes issue ASTERISK-18245)
Reported by: Matt Jordan
(closes issue ASTERISK-18246)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1589/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@345487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change was made because forcegreeting and forcename settings in voicemail could be
circumvented by hanging up after entering a password, because the only way voicemail
currently observes whether a mailbox is new or not is by checking to see if the password
is the same as the mailbox number or not.
(closes issue ASTERISK-18282)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1581/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@345062 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Dummy channels created by ast_dummy_channel_alloc() should be destoyed by
ast_channel_unref(). Using ast_channel_release() needlessly grabs the
channel container lock and can cause a deadlock as a result.
* Analyzed use of ast_dummy_channel_alloc() and made use
ast_channel_unref() when done with the dummy channel. (Primary reason for
the reported deadlock.)
* Made app_dial.c:dial_exec_full() not call ast_call() holding any channel
locks. Chan_local could not perform deadlock avoidance correctly.
(Potential deadlock exposed by this issue. Secondary reason for the
reported deadlock since the held lock was part of the deadlock chain.)
* Fixed some uses of ast_dummy_channel_alloc() not checking the returned
channel pointer for failure.
* Fixed some potential chan=NULL pointer usage in func_odbc.c. Protected
by testing the bogus_chan value.
* Fixed needlessly clearing a 1024 char auto array when setting the first
char to zero is enough in manager.c:action_getvar().
(closes issue ASTERISK-18613)
Reported by: Thomas Arimont
Patches:
jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Thomas Arimont
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes an issue where the voicemail duration was being reported
with a duration significantly less than the actual sound file duration.
Voicemails that contained mostly silence were reporting the duration of
only the sound in the file, as opposed to the duration of the file with
the silence. This patch fixes this by having two durations reported in
the __ast_play_and_record family of functions - the sound_duration and the
actual duration of the file. The sound_duration, which is optional, now
reports the duration of the sound in the file, while the actual full duration
of the file is reported in the duration parameter. This allows the voicemail
applications to use the sound_duration for minimum duration checking, while
reporting the full duration to external parties if the voicemail is kept.
(issue ASTERISK-2234)
(closes issue ASTERISK-16981)
Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1443
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This code change fixes a few issues with the voicemail user override of
emailbody and emailsubject, including escaping the strings, potential memory
leaks, and not overriding the voicemail defaults. Revision 325877 fixed this
for ASTERISK-16795, but did not fix it for ASTERISK-16781. A subsequent
check-in prevented 325877 from being applied to 10. This check-in resolves
both issues, and applies the changes to 1.8, 10, and trunk.
(closes issue ASTERISK-16781)
Reported by: Sebastien Couture
Tested by: mjordan
(closes issue ASTERISK-16795)
Reported by: mdeneen
Tested by: mjordan
Review: https://reviewboard.asterisk.org/r/1374
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333339 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This update adds a new AMI event, TestEvent, which is enabled when the TEST_FRAMEWORK compiler flag is defined. It also adds initial usage of this event to app_voicemail. The TestEvent AMI event is used extensively by the voicemail tests in the Asterisk Test Suite.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Formerly, prepend forwarding would have the user record a message with no useful prompt
and an expectation for the user to push a button on the phone when finished recording.
If a length of silence was detected instead, the recording would be canceled and the user
would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording
would also bug out in the sense that they would write over the original message and get
sent to the recipient regardless of whether they timed out or were accepted. This patch
fixes this issue and adds a prompt which will be played after a timeout informing the
user that they needed to press a button. Currently, the sound files that we have are
somewhat inadquate for this, so after the call we simply have Allison say "Please try
again. Then press pound." which actually relies on two separate sound files. Just one
would be more appropriate.
reporter: Vlad Povorozniuc
Review: https://reviewboard.asterisk.org/r/1327/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@329527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@327852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@326411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@320162 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r312210 | alecdavis | 2011-04-01 21:47:29 +1300 (Fri, 01 Apr 2011) | 29 lines
Merged revisions 312174 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r312174 | alecdavis | 2011-04-01 21:29:49 +1300 (Fri, 01 Apr 2011) | 23 lines
voicemail: get real last_message_index and count_messages, ODBC resequence
change last_message_index to read the max msgnum stored in the database
change count_messages to actually count the number of messages.
last_message_index change:
This fixed overwriting of the last message if msgnum=0 was missing.
Previously every incoming message would overwrite msgnum=1.
count_messages change:
allows us to detect when requencing is required in opneA_mailbox.
resequence enabled for ODBC storage:
Assists with fixing up corrupt databases with gaps, but only when
a user actively opens there mailboxes.
(closes issue #18692,#18582,#19032)
Reported by: elguero
Patches:
based on odbc_resequence_mailbox2.1.diff uploaded by elguero (license 37)
Tested by: elguero, nivek, alecdavis
Review: https://reviewboard.asterisk.org/r/1153/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@312211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r312103 | alecdavis | 2011-04-01 20:25:54 +1300 (Fri, 01 Apr 2011) | 22 lines
Merged revisions 312070 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r312070 | alecdavis | 2011-04-01 19:46:56 +1300 (Fri, 01 Apr 2011) | 16 lines
app_voicemail: close_mailbox needs to respect additional messages while mailbox is open.
close_mailbox leave gaps in message sequence if messages are deleted and new messages
arrive during this time, this is because the shuffle down to slot 0, only shuffles
the number of pre-existing messages when mailbox is opened, ignoring new arrivals.
Fix: in close_mailbox re-evaluate number of messages before the shuffle, this then includes new arrivals.
Happens on filebased or ODBC storage.
(issues #19032,#18582,#18692,#18998)
Reported by: alecdavis,tootai,afosorio
Review: https://reviewboard.asterisk.org/r/1153/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@312117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r306961 | jpeeler | 2011-02-08 13:25:10 -0600 (Tue, 08 Feb 2011) | 15 lines
Merged revisions 306960 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r306960 | jpeeler | 2011-02-08 13:18:50 -0600 (Tue, 08 Feb 2011) | 9 lines
Backup file storing message duration is not used with IMAP_STORAGE, remove code.
The message duration is stored in the body of the email when using IMAP_STORAGE,
so nothing needs to happen with the backup file.
(closes issue #18718)
Reported by: kerframil
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@306962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add and extend the see-also sections to the documentation for applications
and functions in an effort to expand the online documentation of the wiki.
Also check for and update any links to moved documentation in the doc folder.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@304908 65c4cc65-6c06-0410-ace0-fbb531ad65f3
