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981 Commits (9fc52cdaa3e2ba3386f6734b96fe9eeaaa12debb)
Author | SHA1 | Message | Date |
---|---|---|---|
|
c50c68592a |
Patch to play correct sound file when a voicemail's urgent status is removed
We were attempting to play "vm-urgent-removed", which didn't exist. Now we play "vm-marked-nonurgent" which exists and is the correct sound file. Previous behavior was silence and a warning on the CLI. (issue ASTERISK-20280) (closes issue ASTERISK-20280) Reported by: Tomo Takebe Tested by: Rusty Newton Patches: asterisk20280.patch uploaded by Rusty Newton (license 5829) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@376262 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
eb6ad1d9a9 |
Fix error where improper IMAP greetings would be deleted.
(closes issue ASTERISK-20435) Reported by: fhackenberger Patches: asterisk-20435-imap-del-greeting.diff uploaded by Michael L. Young (License #5026) (with suggested modification made by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373735 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
4e5b787aa4 |
Fix VoicemailUserEntry event headers ServerEmail and MailCommand reported values.
The AMI action VoicemailUsersList VoicemailUserEntry event headers ServerEmail and MailCommand did not report the global values if they were not overridden. The VoicemailUserEntry event header ServerEmail was not populated with the global value if the voicemail user did not override it. The VoicemailUserEntry event header MailCommand was never populated with a value. * Removed unused struct ast_vm_user member mailcmd[]. (closes issue AST-973) Reported by: John Bigelow Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372620 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
e6f3d29864 |
Fix memory leaks in app_voicemail when using IMAP storage or realtime config
This patch fixes two memory leaks: 1. When find_user is called with NULL as its first parameter, the voicemail user returned is allocated on the heap. The inboxcount2 function uses find_user in such a fashion when counting new messages, and fails to free the resulting voicemail user object. 2. When populate_defaults is called on a voicemail user, it wipes whatever flags have been set on the object by copying over the global flags object. If the VM_ALLOCED flag was ste on the voicemail user prior to doing so, that flag is removed. This leaks the voicemail user when free_user is later called. (closes issue ASTERISK-19155) Reported by: Filip Jenicek patches: asterisk.patch2 uploaded by Filip Jenicek (license 6277) Patch slightly modified for this commit. Review: https://reviewboard.asterisk.org/r/2096 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372268 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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377caa7fb1 |
Clean up and ensure proper usage of alloca()
This replaces all calls to alloca() with ast_alloca() which calls gcc's __builtin_alloca() to avoid BSD semantics and removes all NULL checks on memory allocated via ast_alloca() and ast_strdupa(). (closes issue ASTERISK-20125) Review: https://reviewboard.asterisk.org/r/2032/ Patch-by: Walter Doekes (wdoekes) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@370642 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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8339cc9684 |
AST-2012-011: Resolve heap corruption issue with voicemail
The heard and deleted arrays in the voicemail state structure were not handled properly following the memory leak fix in r354890 and a fix for an invalid free in r356797. This could result in accessing and writing into freed memory. The allocation for these arrays has been reworked to avoid the possibility of invalid frees, access of freed memory, and crashes that were occurring as a result of this. Locking around accesses and modifications of the voicemail state structure members dh_arraysize, heard, and deleted has been added to prevent simultaneous modification and access when IMAP storage is in use. If IMAP storage is not in use, this locking is not compiled in. Review: https://reviewboard.asterisk.org/r/1994/ (closes issue ASTERISK-19923) Reported by: Dan Delaney Tested by: Dan Delaney, Julian Yap Patches: vm_alloc_fix.diff uploaded by kmoore (license 6273) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@369652 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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0353a57671 |
Fix coverity UNUSED_VALUE findings in core support level files
Most of these were just saving returned values without using them and in some cases the variable being saved to could be removed as well. (issue ASTERISK-19672) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368738 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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6c63f45326 |
Ensure that pages and emails are sent using RFC822-compliant date format
When localization was added to app_voicemail, these headers were altered when they should have remained in en_US format for RFC compliance. This reverts the changes to those two lines. (closes issue ASTERISK-19876) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368520 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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3edf245601 |
Fix a variety of memory leaks
This patch addresses a number of memory leaks in a variety of modules that were found by a static analysis tool. A brief summary of the changes: * app_minivm: free ast_str objects on off nominal paths * app_page: free the ast_dial object if the requested channel technology cannot be appended to the dialing structure * app_queue: if a penalty rule failed to match any existing rule list names, the created rule would not be inserted and its memory would be leaked * app_read: dispose of the created silence detector in the presence of off nominal circumstances * app_voicemail: dispose of an allocated unique ID field for MWI event un-subscribe requests in off nominal paths; dispose of configuration objects when using the secret.conf option * chan_dahdi: dispose of the allocated frame produced by ast_dsp_process * chan_iax2: properly unref peer in CLI command "iax2 unregister" * chan_sip: dispose of the allocated frame produced by sip_rtp_read's call of ast_dsp_process; free memory in parse unit tests * func_dialgroup: properly deref ao2 object grhead in nominal path of dialgroup_read * func_odbc: free resultset in off nominal paths of odbc_read * cli: free match_list in off nominal paths of CLI match completion * config: free comment_buffer/list_buffer when configuration file load is unchanged; free the same buffers any time they were created and config files were processed * data: free XML nodes in various places * enum: free context buffer in off nominal paths * features: free ast_call_feature in off nominal paths of applicationmap config processing * netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct that is allocated by the method. Failures in ast_sockaddr_resolve could result in the users of the method not knowing whether or not the buffer was allocated. The method will now not allocate the ast_sockaddr struct if it will return failure. * pbx: cleanup hash table traversals in off nominal paths; free ignore pattern buffer if it already exists for the specified context * xmldoc: cleanup various nodes when we no longer need them * main/editline: various cleanup of pointers not being freed before being assigned to other memory, cleanup along off nominal paths * menuselect/mxml: cleanup of value buffer for an attribute when that attribute did not specify a value * res_calendar*: responses are allocated via the various *_request method returns and should not be allocated in the various write_event methods; ensure attendee buffer is freed if no data exists in the parsed node; ensure that calendar objects are de-ref'd appropriately * res_jabber: free buffer in off nominal path * res_musiconhold: close the DIR* object in off nominal paths * res_rtp_asterisk: if we run out of ports, close the rtp socket object and free the rtp object * res_srtp: if we fail to create the session in libsrtp, destroy the temporary ast_srtp object (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366880 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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a94fcae21b |
Resolve FORWARD_NULL static analysis warnings
This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding numbers 26, 33, and 29 were already resolved. Those skipped were either extended/deprecated or in areas of code that shouldn't be disturbed. (Closes issue ASTERISK-19650) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366167 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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ae528efea3 |
Coverity Report: Fix issues for error type CHECKED_RETURN for core
(issue ASTERISK-19658) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1905/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366094 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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2c8f87dcb4 |
Support VoiceMail d() option when extension does not exist in channel's context
The VoiceMail d([c]) option is documented to accept digits for a new extension in context <c>, if played during the greeting. This option works fine if the extension being redirected to has an extension with the same initial digit in the channel's current context. If that digit did not happen to exist in some extension, a dialplan match would fail and the user would not be redirected. This patch fixes it such that if the <c> option is used, the extensions are matched in that context as opposed to the caller's original context. (closes issue ASTERISK-18243) Reported by: mjordan Tested by: mjordan Review: https://reviewboard.asterisk.org/r/1892 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365474 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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83d3444284 |
Fix many issues from the NULL_RETURNS Coverity report
Most of the changes here are trivial NULL checks. There are a couple optimizations to remove the need to check for NULL and outboundproxy parsing in chan_sip.c was rewritten to avoid use of strtok. Additionally, a bug was found and fixed with the parsing of outboundproxy when "outboundproxy=," was set. (Closes issue ASTERISK-19654) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365398 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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0da3b6c793 |
Fix handling of negative return code when storing voicemails in ODBC storage
When storing a voicemail message using an ODBC connection to a database, the voicemail message is first stored on disk. The sound file associated with the message is read into memory before being transmitted to the database. When this occurs, a failure in the C library's lseek function would cause a negative value to be passed to the mmap as the size of the memory map to create. This would almost certainly cause the creation of the memory map to fail, resulting in the message being lost. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362201 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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4148e51555 |
Add missing newlines to CLI logging
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361471 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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d3ed07d38a |
Fix crash in app_voicemail during close_mailbox
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 |
13 years ago |
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a37f262426 |
Fix a voicemail memory leak with heard/deleted messages.
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 |
13 years ago |
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b0a70ade4b |
Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352643 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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d78db88681 |
Add regression tests for issue ASTERISK-18838.
Review: https://reviewboard.asterisk.org/r/1572 Reviewed by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347131 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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0d613f777e |
Move setting of voicemail zonetag and locale up a bit.
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 |
14 years ago |
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503d5f8912 |
Guarantee messages go into the right folders with multiple recipients
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 |
14 years ago |
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2fce36ad6b |
Moves voicemail setup password entry to the end of the setup process.
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 |
14 years ago |
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fb6e8a5575 |
Fix previous commit
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341112 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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902b38d21d |
Voicemail compiler flags are 'core' support
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341108 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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f2e1640435 |
Fix deadlock when using dummy channels.
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 |
14 years ago |
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c4cd620d7a |
More silly spacing changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337353 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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6e94c27f6c |
Dumb little spacing fix.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337344 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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f13c3b3fd2 |
Fix for incorrect voicemail duration in external notifications
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 |
14 years ago |
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4b1398a82d |
Make SQL query in app_voicemail.c portable LIMIT is not portable.
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14 years ago |
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92ad64998c |
Fixed improperly formatted TestEvent AMI message in app_voicemail
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333630 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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3a29ee54db |
Fixed incorrect pointer copy to structure copy in revision 333339
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333354 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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792c3a2d56 |
Bug fixes for voicemail user emailsubject / emailbody.
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 |
14 years ago |
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56549c96ab |
Review: https://reviewboard.asterisk.org/r/1364/
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 |
14 years ago |
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3b50c5a387 |
Changes sound file for prepend "then-press-pound" to "vm-then-pound" which is the same
prompt, only it turned out "then-press-pound" was part of extra sounds. Also, vm is more appropriate anyway. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@329529 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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31a1b94622 |
Fixes some voicemail forwarding behavior based around prepend mode.
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 |
14 years ago |
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d4938a111e |
Introduce <support_level> tags in MODULEINFO.
This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328209 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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cafd418c46 |
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@327852 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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9a3fd9a994 |
Removing type attributes, as a change to menuselect makes them no longer necessary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@326469 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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d104b4e701 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@326411 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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40babd5582 |
Patched voicemail user option for emailbody / emailsubject
Incorporated changes per ASTERISK-16795; updated unit tests to check for vmu->emailbody / vmu->emailsubject (closes issue ASTERISK-16795) Reported by: mdeneen Tested by: mjordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@325877 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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ce51fcfb6b |
This patch fixes an issue with using the wrong voicemail folders with greetings.
(closes issue #17871) Reported by: edhorton Patches: digium_bug_17871_2 uploaded by fhackenberger (license 592) Tested by: edhorton, fhackenberger git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@321537 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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164f61d029 |
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@320162 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c23377d8f2 |
Don't create [general] voicemail context when using users.conf
Prior to this patch, app_voicemail would create a [general] context when parsing users.conf. (closes issue #18891) Reported by: pdugas Patches: app_voicemail-ignore-general.patch uploaded by pdugas (license 1222) app_voicemail-ignore-general-style-guidelines.patch uploaded by seanbright (license 71) Tested by: pdugas git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@319367 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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6c3ea80a35 |
Merged revisions 316708 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r316708 | seanbright | 2011-05-04 12:10:59 -0400 (Wed, 04 May 2011) | 15 lines Merged revisions 316707 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r316707 | seanbright | 2011-05-04 12:08:50 -0400 (Wed, 04 May 2011) | 8 lines If sox fails when processing a voicemail, don't delete the original file. (closes issue #18111) Reported by: sysreq Patches: issue18111_trunk.patch uploaded by seanbright (license 71) Tested by: seanbright ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316709 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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a82f1bb995 |
Fix a bunch of compiler warnings generated by gcc 4.6.0.
Most of these are -Wunused-but-set-variable, but there were a few others mixed in here, as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316265 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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8fe6967f1d |
app_voicemail: close_mailbox change LOG_WARNING to LOG_NOTICE
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@313002 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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62e679f784 |
Merged revisions 312210 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ 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 ........ 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/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@312211 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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83aeb52dd0 |
Merged revisions 312103 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ 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 ........ 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/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@312117 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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0a186e3f4f |
Cross-reference VoiceMail() and VoiceMailMain() in the xml docs.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@311751 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Merged revisions 310141 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r310141 | tilghman | 2011-03-09 23:51:37 -0600 (Wed, 09 Mar 2011) | 12 lines Merged revisions 310140 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r310140 | tilghman | 2011-03-09 23:38:44 -0600 (Wed, 09 Mar 2011) | 5 lines Initialize column size to 0 to deal with a potential UnixODBC bug on 64-bit systems. (closes issue #18295) Reported by: pruiz ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@310142 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |